Sound field correction device, sound field correction method, and program

Abstract

To effectively suppress sound lag among multiple speakers.SOLUTION: A sound field correction device includes: a test sound output unit that outputs a first test sound from a first speaker and a second speaker, outputs a second test sound from the first speaker when a first waiting time has elapsed after the output of the first test sound, and outputs the second test sound from the second speaker when a second waiting time has elapsed after the output of the first test sound; and a calculation unit that calculates a first time difference between the timing at which the first test sound output from the first speaker reaches a microphone and the timing at which the first test sound output from the second speaker reaches the microphone and a second time difference between the timing at which the second test sound output from the first speaker reaches the microphone and the timing at which the second test sound output from the second speaker reaches the microphone, on the basis of a composite sound collected and recorded by the microphone.SELECTED DRAWING: Figure 2

Description

Technical Field

[0001] Embodiments of the present invention relate to a sound field correction device, a sound field correction method, and a program.

Background Art

[0002] In an acoustic device that outputs sound from a plurality of speakers, there may be a sound shift due to the difference in the timing at which the sound output from each speaker reaches the listening position. As a technique for solving such a problem, there is a technique of picking up the test sound output from the speaker with a microphone, calculating the distance between the speaker and the microphone based on the output signal of the microphone, and adjusting the sound output timing according to the distance.

Prior Art Documents

Patent Documents

[0003]

Patent Document 1

Summary of the Invention

Problems to be Solved by the Invention

[0004] The above-mentioned sound shift occurs due to not only the distance difference between each speaker and the listening position, but also various factors such as communication delay due to the intervention of wireless communication and data processing time in the acoustic device. Depending on the prior art, the sound shift caused by such various factors may not be sufficiently suppressed.

[0005] The problem to be solved by the embodiments of the present invention is to provide a sound field correction device, a sound field correction method, and a program capable of effectively suppressing the sound shift between a plurality of speakers.

Means for Solving the Problems

[0006] One embodiment of the present invention is an acoustic field correction device that performs processing for reducing the deviation in the timing at which sound output from each of a plurality of speakers reaches a predetermined listening position. The device causes a first test sound of a first frequency to be output from a first speaker and a second speaker, and when a first waiting time has elapsed after the output of the first test sound, causes a second test sound of a second frequency different from the first frequency to be output from the first speaker. When a second waiting time different from the first waiting time has elapsed after the output of the first test sound, a test sound output unit that causes the second test sound to be output from the second speaker, a recording unit that records a composite sound including the first test sound and the second test sound collected by a microphone arranged at the listening position and output from each of the first speaker and the second speaker, and based on the result of a frequency analysis process capable of detecting components corresponding to the first frequency and components corresponding to the second frequency from the recorded composite sound, the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone A calculation unit that calculates a first time difference that is the time difference between, and the timing at which the second test sound output from the first speaker reaches the microphone and the timing at which the second test sound output from the second speaker reaches the microphone A correction unit that corrects at least one of the output timing of the sound from the first speaker or the output timing of the sound from the second speaker based on the second time difference, which is the time difference between the two.

Brief Description of the Drawings

[0007]

Figure 1

Figure 2

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Figure 9

[0008] Hereinafter, exemplary embodiments of the present invention will be disclosed.

[0009] FIG. 1 is a block diagram showing an example of the hardware configuration of the audio device 1 according to the embodiment. The audio device 1 is a device capable of outputting sound from a plurality of speakers, and can be, for example, a stereo, a video playback device, a recording device, a television, a home theater system, or the like.

[0010] The audio device 1 according to the present embodiment includes a sound field correction device 5, an internal speaker 31A, an external connection I / F (Interface) 32, and a remote control 41, and can be connected to an external speaker 31B via the external connection I / F 32. The internal speaker 31A is an example of a first speaker, and the external speaker 31B is an example of a second speaker.

[0011] The sound field correction device 5 performs a sound field correction process for adjusting the output timing of the voice AS1 output from the internal speaker 31A and the output timing of the voice AS2 output from the external speaker 31B so that no voice shift occurs. The sound field correction process is executed so as to reduce the shift between the timing at which the voice AS1 output from the internal speaker 31A reaches the user's listening position and the timing at which the voice AS2 output from the external speaker 31B reaches the listening position. The listening position in the present embodiment is assumed to be the position where the remote controller 41 exists.

[0012] The sound field correction device 5 according to the present embodiment includes a CPU (Central Processing Unit) 11, a memory 12, a storage 13, a user I / F 14, a communication I / F 15, a voice decoder 21, a voice input ADC (Analog to Digital Converter) 22, a DSP (Digital Signal Processor) 23, a first delay circuit 25A, and a second delay circuit 25B, and these components are communicably connected to each other via a communication bus 20.

[0013] The CPU 11 executes predetermined arithmetic processing and control processing in accordance with programs (including firmware, application software, etc.) stored in the memory 12 and the like. The memory 12 is a main storage device including a RAM (Random Access Memory), a ROM (Read Only Memory), etc., and functions as a storage area for programs, a work area for the CPU 11, and the like. The storage 13 is an auxiliary storage device including a non-volatile memory such as an SSD (Solid State Drive) or an HDD (Hard Disk Drive), and enables writing and reading of various data. The user I / F 14 is a device that enables reception of inputs from the user, output of information to the user, etc., and can be, for example, a display, an input button, etc. The communication I / F 15 is a device that enables communication with other electronic devices connected via a predetermined communication network. The communication I / F 15 of the present embodiment establishes wireless communication conforming to a predetermined standard with the remote controller 41.

[0014] The audio decoder 21 is a device that converts audio data recorded on a predetermined medium (such as a CD, DVD, Blu-ray (registered trademark) disc, removable media, etc.), audio data included in a broadcast wave, or audio data obtained from a network such as a CSP (Communications Service Provider) into a digital signal in a format that can be output from the internal speaker 31A and the external speaker 31B. The audio input ADC 22 is a device that converts an analog signal of audio input from an external device into a digital signal.

[0015] The DSP 23 is a processor that executes predetermined processing on the digital signal corresponding to the sound output from the internal speaker 31A and the external speaker 31B, and generates audio signals of the audio AS1 and AS2 to be listened to, and audio signals of the first test sound TS1 and the second test sound TS2 described later.

[0016] The first delay circuit 25A is a circuit that delays the output timing of the sound output from the internal speaker 31A according to a correction signal (delay signal) output from the CPU 11. The second delay circuit 25B is a circuit that delays the output timing of the sound output from the external speaker 31B according to the correction signal output from the CPU 11.

[0017] The external connection I / F 32 transmits the audio signal generated by the DSP 23 and whose output timing is adjusted by the second delay circuit 25B to the external speaker 31B via a predetermined communication network. The communication network is not particularly limited, and can be, for example, wireless communication conforming to a predetermined standard, wired communication using a predetermined cable, etc. The predetermined standard can be, for example, Bluetooth (registered trademark), Wi-Fi (registered trademark), etc.

[0018] The remote controller 41 is a device operable by a user who wishes to listen to the voice AS1, AS2, and includes a microphone 45, a wireless modulation circuit 46, and a transmitter 47. The microphone 45 is a device that converts the collected sound into an electrical signal (analog signal). The wireless modulation circuit 46 is a circuit that modulates the electrical signal generated by the microphone 45 into a signal (digital signal) in a form capable of wireless communication conforming to a predetermined standard. The transmitter 47 is a device that transmits the signal modulated by the wireless modulation circuit 46 to the sound field correction device 5. In addition to the above, the remote controller 41 is provided with buttons and the like for receiving operations by the user, but the description thereof is omitted here. Further, in the present embodiment, a configuration in which the microphone 45 is provided in the remote controller 41 is illustrated, but the microphone 45 may be an independent device.

[0019] The sound field correction device 5 according to the present embodiment outputs a first test sound TS1 and a second test sound TS2 from the internal speaker 31A and the external speaker 31B, respectively, when executing the sound field correction process. The microphone 45 mounted on the remote controller 41 picks up a composite sound including the first test sound TS1 and the second test sound TS2, and the transmitter 47 transmits the acoustic signal St of the composite sound picked up by the microphone 45 to the sound field correction device 5. The sound field correction device 5 records the acoustic signal St received from the remote controller 41 and executes a frequency analysis process on the recorded acoustic signal St. The sound field correction device 5 calculates (estimates) the deviation in the timing at which the voices AS1, AS2 output from the internal speaker 31A and the external speaker 31B reach the listening position based on the result of the frequency analysis process, and corrects at least one of the output timing of the voice AS1 from the internal speaker 31A or the output timing of the voice AS2 from the external speaker 31B so as to reduce the deviation.

[0020] FIG. 2 is a block diagram showing an example of the functional configuration of the sound field correction device 5 according to the embodiment. The sound field correction device 5 according to the present embodiment includes a voice output unit 101, a test sound output unit 102, a reception unit 103, a recording unit 104, an arithmetic unit 105, and a correction unit 106. These functional units 101 to 106 can be realized by the cooperation of hardware and software (program) as exemplified in FIG. 1. Further, at least a part of these functional units 101 to 106 may be realized by dedicated hardware (such as a circuit).

[0021] The voice output unit 101 causes the internal speaker 31A and the external speaker 31B to output the voices AS1 and AS2 to be listened to, respectively.

[0022] The test sound output unit 102 causes the internal speaker 31A and the external speaker 31B to output the first test sound TS1 and the second test sound TS2, respectively, when performing the sound field correction process. At this time, the frequency of the first test sound TS1 and the frequency of the second test sound TS2 are different from each other. Further, the time interval (first standby time) between the first test sound TS1 and the second test sound TS2 output from the internal speaker 31A and the time interval (second standby time) between the first test sound TS1 and the second test sound TS2 output from the external speaker 31B are different from each other.

[0023] FIG. 3 is a diagram for explaining an example of the characteristics of the first test sound TS1 and the second test sound TS2 according to the embodiment. As illustrated in FIG. 3, the first test sound TS1 is a sound having a first frequency Ft1 (for example, about 5000 Hz), and the second test sound TS2 is a sound having a second frequency Ft2 (for example, about 2000 Hz) different from the first frequency Ft1. In this example, the output time Δt of the first test sound TS1 and the output time Δt of the second test sound TS2 are the same, but the present invention is not limited thereto. The specific numerical value of the output time Δt is not particularly limited, but may be, for example, about 50 ms.

[0024] In the internal speaker 31A, after the output of the first test sound TS1, the second test sound TS2 is output when the first waiting time Δtw1 has elapsed. In the external speaker 31B, after the output of the first test sound TS1, the second test sound TS2 is output when the second waiting time Δtw2 different from the first waiting time Δtw1 has elapsed. The specific numerical values of the first waiting time Δtw1 and the second waiting time Δtw2 are not particularly limited. For example, the first waiting time Δtw1 is about 50 ms, and the difference between the first waiting time Δtw1 and the second waiting time Δtw2 can be about 20 ms.

[0025] The output of the first test sound TS1 and the second test sound TS2 as described above can be realized, for example, by the following operation of the test sound output unit 102. First, the test sound output unit 102 simultaneously outputs a first instruction signal for causing the internal speaker 31A and the external speaker 31B to output the first test sound TS1. After that, after the end of the output of the first instruction signal, when the first waiting time Δtw1 has elapsed, the test sound output unit 102 outputs a second instruction signal for causing the internal speaker 31A to output the second test sound TS2. Also, after the end of the output of the first instruction signal, when the second waiting time Δtw2 has elapsed, the test sound output unit 102 outputs a third instruction signal for causing the external speaker 31B to output the second test sound TS2.

[0026] Returning to FIG. 2, the receiving unit 103 receives the acoustic signal St of the composite sound that includes the first test sound TS1 and the second test sound TS2 as described above and is picked up by the microphone 45 arranged at the listening position. The receiving unit 103 of the present embodiment receives the acoustic signal St transmitted from the remote controller 41.

[0027] The recording unit 104 records the acoustic signal St received by the receiving unit 103 in a predetermined storage device (for example, the storage 13 or the like).

[0028] The arithmetic unit 105 performs a frequency analysis process capable of detecting components corresponding to the first frequency Ft1 (for example, frequency components within a predetermined error range from 5000 Hz) and the second frequency Ft2 (for example, frequency components within a predetermined error range from 2000 Hz) included in the acoustic signal St with respect to the acoustic signal St recorded by the recording unit 104. The specific method of the frequency analysis process should not be particularly limited, and for example, FFT (Fast Fourier Transform) or the like can be used. Based on the result of the frequency analysis process, the arithmetic unit 105 calculates a first time difference Δt1 and a second time difference Δt2. The first time difference Δt1 is the time difference between the timing when the first test sound TS1 output from the internal speaker 31A reaches the microphone 45 and the timing when the first test sound TS1 output from the external speaker 31B reaches the microphone 45. The second time difference Δt2 is the time difference between the timing when the second test sound TS2 output from the internal speaker 31A reaches the microphone 45 and the timing when the second test sound TS2 output from the external speaker 31B reaches the microphone 45.

[0029] The correction unit 106 corrects at least one of the output timing of the voice AS1 from the internal speaker 31A or the output timing of the voice AS2 from the external speaker 31B based on the first time difference Δt1 and the second time difference Δt2 calculated by the arithmetic unit 105. At this time, the correction unit 106 of the present embodiment determines the order relationship between the timing when the first test sound TS1 output from the internal speaker 31A reaches the microphone 45 and the timing when the first test sound TS1 output from the external speaker 31B reaches the microphone 45, that is, the order relationship of the delay between the internal speaker 31A and the external speaker 31B, based on the magnitude relationship between the first time difference Δt1 and the second time difference Δt2. Then, based on the order relationship, the correction unit 106 relatively delays either the output timing of the voice AS1 of the internal speaker 31A or the output timing of the voice AS2 of the external speaker 31B so that the first time difference Δt1 becomes smaller.

[0030] With the above configuration, it is possible to effectively suppress voice deviation at the listening position caused by various factors such as the distance difference between each speaker 31A, 31B and the listening position, communication delay due to the intervention of wireless communication, and data processing time in the audio device.

[0031] FIG. 4 is a diagram showing an example of frequency analysis processing for the acoustic signal St of the composite sound according to the embodiment. In the graph illustrated at the upper part of FIG. 4, the horizontal axis (Time) indicates the elapsed time from a predetermined reference time (for example, the output time of an instruction signal (first instruction signal) for outputting the first test sound TS1, the start time of recording of the composite sound, etc.), and the vertical axis (Intensity) indicates the signal intensity of the composite sound picked up by the microphone 45. The graph illustrated at the lower part of FIG. 4 shows the analysis result F when frequency analysis processing is performed on the acoustic signal St. The horizontal axis (Frequency) corresponds to the frequency components included in the composite sound, and the vertical axis (Power) corresponds to the power spectrum. Here, the analysis result F in the case of performing FFT processing on the time window W for a predetermined time frame set in the acoustic signal St is illustrated.

[0032] In the present embodiment, a power value corresponding to a predetermined target frequency range R is calculated from the analysis result F. The target frequency range R is set with reference to the first frequency Ft1 when detecting (extracting) the first frequency Ft1, and is set with reference to the second frequency Ft2 when detecting the second frequency Ft2. For example, the target frequency range R at the time of detecting the first frequency Ft1 is set as (Ft1 ± x%). For example, when Ft1 = 5000 (Hz) and x = 5, R = 5000 ± 250 (Hz). Similarly, the target frequency range R at the time of extracting the second frequency Ft2 is set as (Ft2 ± x%). For example, when Ft2 = 2000 (Hz) and x = 20, R = 2000 ± 400 (Hz). By the above-described processing, for each of a plurality of time windows W that are gradually shifted at a predetermined time interval, the power value corresponding to the first frequency Ft1 (for example, R = 5000 ± 250) and the power value corresponding to the second frequency Ft2 (R = 2000 ± 400) are obtained.

[0033] FIG. 5 is a diagram showing an example of the detection result F1 of the first frequency Ft1 according to the embodiment. FIG. 6 is a diagram showing an example of the detection result F2 of the second frequency Ft2 according to the embodiment. In the detection result F1, the horizontal axis corresponds to time, and the vertical axis corresponds to the power value corresponding to the target frequency range R (for example, 5000 ± 250) of the first frequency Ft1. In the detection result F2, the horizontal axis corresponds to time, and the vertical axis corresponds to the power value corresponding to the target frequency range R (for example, 2000 ± 400) of the second frequency Ft2. The time axis of the detection result F1 and the time axis of the detection result F2 are the same.

[0034] In FIG. 5, the first rising point tr11, the first falling point tf11, the second rising point tr12, and the second falling point tf12 corresponding to the first frequency Ft1 are illustrated. The first rising point tr11 is the first time point when the power value corresponding to the target frequency range R of the first frequency Ft1 changes from a state smaller than the threshold Th1 to a state equal to or greater than the threshold Th1. The first falling point tf11 is the first time point when the power value corresponding to the target frequency range R of the first frequency Ft1 changes from a state equal to or greater than the threshold Th1 to a state smaller than the threshold Th1. The second rising point tr12 is the second time point when the power value corresponding to the target frequency range R of the first frequency Ft1 changes from a state smaller than the threshold Th1 to a state equal to or greater than the threshold Th1. The second falling point tf12 is the second time point when the power value corresponding to the target frequency range R of the first frequency Ft1 changes from a state equal to or greater than the threshold Th1 to a state smaller than the threshold Th1.

[0035] Here, it is assumed that the first test sound TS1 output from the internal speaker 31A reaches the microphone 45 before the first test sound TS1 output from the external speaker 31B. In this case, the first rising point tr11 is the time point when the first test sound TS1 output from the internal speaker 31A reaches the microphone 45, and the second rising point tr12 is the time point when the first test sound TS1 output from the external speaker 31B reaches the microphone 45. Therefore, the time from the first rising point tr11 to the second rising point tr12 becomes the first time difference Δt1.

[0036] In FIG. 6, the third rising time point tr21, the third falling time point tf21, the fourth rising time point tr22, and the fourth falling time point tf22 corresponding to the second frequency Ft2 are illustrated. The third rising time point tr21 is the first time point at which the power value corresponding to the target frequency range R of the second frequency Ft2 changes from a state smaller than the threshold Th2 to a state equal to or greater than the threshold Th2. The third falling time point tf21 is the first time point at which the power value corresponding to the target frequency range R of the second frequency Ft2 changes from a state equal to or greater than the threshold Th2 to a state smaller than the threshold Th2. The fourth rising time point tr22 is the second time point at which the power value corresponding to the target frequency range R of the second frequency Ft2 changes from a state smaller than the threshold Th2 to a state equal to or greater than the threshold Th2. The fourth falling time point tf22 is the second time point at which the power value corresponding to the target frequency range R of the second frequency Ft2 changes from a state equal to or greater than the threshold Th2 to a state smaller than the threshold Th2.

[0037] Here, it is assumed that the second test sound TS2 output from the internal speaker 31A reaches the microphone 45 before the second test sound TS2 output from the external speaker 31B. In this case, the third rising time point tr21 is the time point at which the second test sound TS2 output from the internal speaker 31A reaches the microphone 45, and the fourth rising time point tr22 is the time point at which the second test sound TS2 output from the external speaker 31B reaches the microphone 45. Therefore, the time from the third rising time point tr21 to the fourth rising time point tr22 becomes the second time difference Δt2.

[0038] In the above, as illustrated in FIG. 5, the case where the first falling time point tf11 exists before the second rising time point tr12 on the time axis has been described. In such a case, as described above, the first time difference Δt1 can be calculated based on the first rising time point tr11 and the second rising time point tr12. However, when the first time difference Δt1 is smaller than the output time Δt of the first test sound TS1 (see FIG. 3), the first wave W1 corresponding to the first test sound TS1 that reached the microphone 45 first and the second wave W2 corresponding to the first test sound TS1 that reached the microphone 45 later overlap, and the first falling time point tf11 and the second rising time point tr12 do not appear. Hereinafter, a method for calculating the first time difference Δt1 in such a case will be described.

[0039] FIG. 7 is a diagram for explaining an example of a method for calculating the first time difference Δt1 when the first time difference Δt1 according to the embodiment is smaller than the output time Δt of the first test sound TS1. At this time, as shown in FIG. 7, since the first wave W1 and the second wave W2 overlap, only the first rising time point tr11 and the second falling time point tf12 appear on the time axis. In such a case, the first time difference Δt1 can be calculated by subtracting the output time Δt of the first test sound TS1 from the time Δt' from the first rising time point tr11 to the second falling time point tf12.

[0040] The above calculation method can be similarly applied to the second time difference Δt2. That is, when the second time difference Δt2 is smaller than the output time Δt of the second test sound TS2, the second time difference Δt2 can be calculated by subtracting the output time Δt of the second test sound TS2 from the time from the third rising time point tr21 to the fourth falling time point tf22.

[0041] Also, as described above, the correction unit 106 of the present embodiment determines the priority relationship of the delay between the internal speaker 31A and the external speaker 31B based on the magnitude relationship between the first time difference Δt1 and the second time difference Δt2.

[0042] FIG. 8 is a diagram showing an example of a method for determining the precedence relationship of delays according to an embodiment. In FIG. 8, the relationship between the magnitude relationship between the first time difference Δt1 and the second time difference Δt2, the precedence relationship of the delays, and the speaker to be delay-controlled is illustrated. In FIG. 8, the value a represents the difference between the first standby time Δtw1 and the second standby time Δtw2 (a = Δtw2 - Δtw1).

[0043] As shown in FIG. 8, when the first time difference Δt1 is greater than or equal to the second time difference Δt2 (Δt1 ≧ Δt2), it is determined that the first test sound TS1 from the external speaker 31B reaches the microphone 45 earlier than the first test sound TS1 from the internal speaker 31A. At this time, the correction unit 106 delays the output timing of the sound AS2 of the external speaker 31B.

[0044] When the first time difference Δt1 is smaller than the second time difference Δt2 and the first time difference Δt1 is smaller than a / 2 (Δt1 < Δt2 and Δt1 < a / 2), it is determined that the first test sound TS1 from the external speaker 31B reaches the microphone 45 earlier than the first test sound TS1 from the internal speaker 31A. At this time, the correction unit 106 delays the output timing of the sound AS2 of the external speaker 31B.

[0045] When the first time difference Δt1 is smaller than the second time difference Δt2 and the first time difference Δt1 is greater than or equal to a / 2 (Δt1 < Δt2 and Δt1 ≧ a / 2), it is determined that the first test sound TS1 from the internal speaker 31A reaches the microphone 45 earlier than the first test sound TS1 from the external speaker 31B. At this time, the correction unit 106 delays the output timing of the sound AS1 of the internal speaker 31A.

[0046] Note that the above determination method is an example, and the method for determining the precedence relationship of delays is not limited to the above.

[0047] FIG. 9 is a flowchart showing an example of processing in the sound field correction apparatus 5 according to the embodiment. When the sound field correction processing starts, the test sound output unit 102 simultaneously outputs a first instruction signal for causing the internal speaker 31A and the external speaker 31B to output a first test sound TS1 (S101). Thereafter, after the output of the first instruction signal ends, the test sound output unit 102 determines whether or not a first waiting time Δtw1 has elapsed (S102). If the first waiting time Δtw1 has not elapsed (S102: No), the determination in step S102 is continued. When the first waiting time Δtw1 has elapsed (S102: Yes), the test sound output unit 102 outputs a second instruction signal for causing the internal speaker 31A to output a second test sound TS2 (S103). Thereafter, after the output of the first instruction signal ends, the test sound output unit 102 determines whether or not a second waiting time Δtw2 has elapsed (S104). If the second waiting time Δtw2 has not elapsed (S104: No), the determination in step S104 is continued. When the second waiting time Δtw2 has elapsed (S104: Yes), the test sound output unit 102 outputs a second instruction signal for causing the external speaker 31B to output a second test sound TS2 (S105). By the above processing, the first test sound TS1 and the second test sound TS2 are output from the internal speaker 31A with a first waiting time Δtw1 therebetween, and the first test sound TS1 and the second test sound TS2 are output from the external speaker 31B with a second waiting time Δtw2 therebetween.

[0048] The composite sound including the first test sound TS1 and the second test sound TS2 output from each of the internal speaker 31A and the external speaker 31B is picked up by the microphone 45 arranged at the listening position, and the picked-up composite sound is recorded (S106). Then, the arithmetic unit 105 detects a first frequency component corresponding to the first test sound TS1 and a second frequency component corresponding to the second test sound TS2 by performing a frequency analysis process on the recorded composite sound (S107), and calculates a first time difference Δt1 and a second time difference Δt2 based on the detection results (S108). As described above, the first time difference Δt1 is the time difference between the timing when the first test sound TS1 output from the internal speaker 31A reaches the microphone 45 and the timing when the first test sound TS1 output from the external speaker 31B reaches the microphone 45, and the second time difference Δt2 is the time difference between the timing when the second test sound TS2 output from the internal speaker 31A reaches the microphone 45 and the timing when the second test sound TS2 output from the external speaker 31B reaches the microphone 45.

[0049] Thereafter, the correction unit 106 determines the delay amount and the order of the delay between the internal speaker 31A and the external speaker 31B based on the first time difference Δt1 and the second time difference Δt2 calculated as described above (S109). Then, the correction unit 106 corrects at least one of the output timing of the audio AS1 from the internal speaker 31A or the output timing of the audio AS2 from the external speaker 31B so that the delay amount is reduced based on the determination result (S110).

[0050] According to the above embodiment, the composite sound including the first test sound TS1 and the second test sound TS2 output from each of the two speakers (the internal speaker 31A and the external speaker 31B in this embodiment) is recorded, the audio delay between the two speakers 31A and 31B is detected based on the analysis result of the recorded composite sound, and the output timings of the audio AS1 and AS2 from the two speakers 31A and 31B are corrected so that the delay is reduced. Thereby, the audio shift at the listening position caused by various factors can be effectively suppressed.

[0051] Also, according to the present embodiment, two types of test sounds including a first test sound TS1 and a second test sound TS2, and having different waiting times Δtw1 and Δtw2 between the first test sound TS1 and the second test sound TS2 are used to detect a delay between two speakers 31A and 31B. By such a method, it is not necessary to output test sounds of different frequencies from each of the two speakers 31A and 31B, so that the mechanism for generating the test sounds can be simplified.

[0052] In the above embodiment, the case of correcting the voice shift between the internal speaker 31A and the external speaker 31B is exemplified, but the number and type of speakers are not limited to this. For example, correction between two internal speakers or correction between two external speakers can also be performed by the same method as described above. Further, when there are three or more speakers, the voice shift between all speakers can be reduced by sequentially performing the same processing as described above for each pair of two of them.

[0053] A program for realizing the functions of the sound field correction device 5 as described above may be recorded and provided on a computer-readable recording medium such as a CD-ROM, a flexible disk (FD), a CD-R, a DVD (Digital Versatile Disk), etc. in a form installable or executable on a computer. Further, the program may be stored on a computer connected to a network such as the Internet and provided by being downloaded via the network. Further, the program may be configured to be provided or distributed via a network such as the Internet.

[0054] As described above, the embodiments of the present invention have been explained. However, these embodiments are presented as examples and are not intended to limit the scope of the invention. These novel embodiments can be implemented in various other forms, and various omissions, replacements, and changes can be made without departing from the gist of the invention. These embodiments and their modifications are included in the scope and gist of the invention, and are also included in the invention described in the claims and the equivalent scope thereof.

Explanation of Reference Numerals

[0055] 1... Acoustic device, 5... Sound field correction device, 11... CPU, 12... Memory, 13... Storage, 14... User I / F, 15... Communication I / F, 20... Communication bus, 21... Audio decoder, 22... Audio input ADC, 23... DSP, 25A... First delay circuit, 25B... Second delay circuit, 31A... Internal speaker, 31B... External speaker, 41... Remote control, 45... Microphone, 46... Wireless modulation circuit, 47... Transmitter, 101... Audio output section, 102... Test tone output section, 103... Receiver, 104... Recording section, 105... Arithmetic section, 106... Correction section, AS1, AS2... Audio, R... Target frequency range, St... Acoustic signal, tr11... First rising time point, tf11... First falling time point, tr12... Second rising time point, tf12... Second falling time point, tr21... Third rising time point, tf21... Third falling time point, tr22... Fourth rising time point, tf22... Fourth falling time point, TS1... First test tone, TS2... Second test tone, W... Time window, Δt... Output time, Δt1... First time difference, Δt2... Second time difference, Δtw1... First standby time, Δtw2... Second standby time

Claims

1. An acoustic field correction device that performs processing for reducing a deviation in the timing at which sound output from each of a plurality of speakers reaches a predetermined listening position, a test sound output unit that outputs a first test sound of a first frequency from a first speaker and a second speaker, and outputs a second test sound of a second frequency different from the first frequency from the first speaker when a first waiting time has elapsed after the output of the first test sound, and outputs the second test sound from the second speaker when a second waiting time different from the first waiting time has elapsed after the output of the first test sound; a recording unit that records a composite sound including the first test sound and the second test sound output from each of the first speaker and the second speaker, the composite sound being picked up by a microphone disposed at the listening position; an arithmetic unit that calculates a first time difference, which is a time difference between the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone, and a second time difference, which is a time difference between the timing at which the second test sound output from the first speaker reaches the microphone and the timing at which the second test sound output from the second speaker reaches the microphone, based on a result of a frequency analysis process capable of detecting a component corresponding to the first frequency and a component corresponding to the second frequency from the recorded composite sound; a correction unit that determines a temporal relationship between the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone based on a magnitude relationship between the first time difference and the second time difference, and relatively delays either the output timing of the sound of the first speaker or the output timing of the sound of the second speaker so that the first time difference becomes smaller based on the temporal relationship; An acoustic field correction device comprising the above components.

2. An acoustic field correction method for reducing a deviation in the timing at which sound output from each of a plurality of speakers reaches a predetermined listening position, a step of outputting a first test sound of a first frequency from a first speaker and a second speaker; a step of outputting a second test sound having a second frequency different from the first frequency from the first speaker when a first waiting time has elapsed after the output of the first test sound; a step of outputting the second test sound from the second speaker when a second waiting time different from the first waiting time has elapsed after the output of the first test sound; a step of recording, by a microphone disposed at the listening position, a composite sound including the first test sound and the second test sound output from each of the first speaker and the second speaker; Based on the result of a frequency analysis process capable of detecting a component corresponding to the first frequency and a component corresponding to the second frequency from the recorded composite sound, the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone, a first time difference which is the time difference therebetween, and the timing at which the second test sound output from the first speaker reaches the microphone and the timing at which the second test sound output from the second speaker reaches the microphone, a second time difference which is the time difference therebetween, are calculated; Based on the magnitude relationship between the first time difference and the second time difference, the temporal relationship between the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone is determined, and based on the temporal relationship, either the output timing of the sound of the first speaker or the output timing of the sound of the second speaker is relatively delayed so that the first time difference becomes smaller; A sound field correction method including the above.

3. In an information processing apparatus that performs processing for reducing a deviation in the timing at which sound output from each of a plurality of speakers reaches a predetermined listening position, a process of outputting a first test sound having a first frequency from a first speaker and a second speaker; a process of outputting a second test sound having a second frequency different from the first frequency from the first speaker when a first waiting time has elapsed after the output of the first test sound; a process of outputting the second test sound from the second speaker when a second waiting time different from the first waiting time has elapsed after the output of the first test sound; A process of recording a composite sound that is picked up by a microphone disposed in the listening position and includes the first test sound and the second test sound output from each of the first speaker and the second speaker; Based on the results of a frequency analysis process capable of detecting a component corresponding to the first frequency and a component corresponding to the second frequency from the recorded composite sound, a first time difference that is the time difference between the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone, and a second time difference that is the time difference between the timing at which the second test sound output from the first speaker reaches the microphone and the timing at which the second test sound output from the second speaker reaches the microphone are calculated; Based on the magnitude relationship between the first time difference and the second time difference, the order relationship between the timing at which the first test sound output from the first speaker reaches the microphone and the timing at which the first test sound output from the second speaker reaches the microphone is determined, and based on the order relationship, a process of relatively delaying either the output timing of the sound of the first speaker or the output timing of the sound of the second speaker so that the first time difference becomes smaller; A program for executing the above.

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